In the last several years, tremendous interest has developed in the “IP telephony” technological area. IP telephony generally involves the transmission of telephone calls through packet-switched data networks employing the Internet Protocol (IP), which is utilized to send data over the Internet. When referring to an IP network, those skilled in the art generally refer to their Local Area Network (LAN) and/or Wide Area Network (WAN) as networks that can communicate with the Internet. The term Internet Protocol and its acronym IP are well known in the telecommunications arts and generally refers to a telecommunications standard covering software and algorithms thereof that keep track of internet addresses for different nodes, routes outgoing messages, and recognizes incoming messages. IP permits a packet to traverse multiple networks on the way to its final destination.
In order to take advantage of the inherent flexibility of IP networks, such as the Internet, IP telephones have been developed, which are configured with IP telephony features. From an end user perspective, the IP telephone typically appears similar to a standard telephone. Most IP telephones include a handset that the user can pick up, speak into, and listen through. IP telephones also typically posses a push button numerical keypad for dialing. When a caller dials a number, the IP telephone signals which keys have been activated and sends this information through the LAN/WAN. IP telephones are different from traditional phones in that they can directly connect to the LAN/WAN in the same manner as, for example, a desktop computer. With most IP telephones, the desktop computer data line can then connect directly to a hub in the IP telephone for network access. Thus, the advantage of an IP telephone is that only a single Ethernet line is required for providing both phone communications and computer network access.
Currently, the common approach to IP telephony involves the use of a speech encoder to compress 8 kHz sampled speech to a low bit rate, package the compressed bit-stream into packets, and then transmit the packets over IP networks. At the receiving end, the compressed bit-stream can be extracted from the received packets, and a speech decoder can be used to decode the compressed bit-stream back to 8 kHz-sampled speech. The term “codec” (coder and decoder) is commonly used to denote the combination of the encoder and the decoder. The current generation of IP telephony products typically uses existing speech codecs that were designed to compress 8 kHz telephone speech to very low bit rates. Examples of such codecs include the ITU-T G.723.1 at 6.3 kb/s, G.729 at 8 kb/s, and G.729A at 8 kb/s. All of these codecs have somewhat degraded speech quality when compared with the ITU-T 64 kb/s G.711 PCM and, of course, they all still have the same 300 to 3,400 Hz bandwidth limitation. IP telephones currently operate using traditional G.711 64 kbps codecs. These 64 kbps samples are grouped together at the IP Endpoint (i.e., phone or CPE gateway) and sent initially over a LAN and a WAN and thereafter to an IP gateway.
Note that the acronym CPE generally refers to “Customer Provided Equipment” or “Customer Premises Equipment.” At the IP gateway the individual 64 kbps samples are unpacked and sent over a GR-303 or V5 interface to a class 5-circuit switch as one 64 kbps DS0. Note that the term “GR-303” refers to a well-known telecommunications standard covering high-level control interfaces to dumb switches. GR-303 comprises generic requirements for next-generation integrated digital loop carrier systems. The acronym “DS” generally refers to “Digital Signal,” and is commonly utilized in the context of DS0, DS1, DS2, etc. Thus, “digital signal X” or DSX is a term which refers to the series of standard digital transmission rates or levels based on DS0, a transmission rate of 64 Kbps, the bandwidth normally used for one telephone voice channel. Both the North American T-carrier system and the European E-carrier systems of transmission operate using the DS series as a base multiple. The digital signal is carried inside the carrier system. DS0 is the base for the digital signal X series.
Wide-band IP telephones are currently being developed for use with IP PBXs. An IP PBX can switch these wide-band 256 kbps voice signals from one IP telephone to another offering broader frequency response and dynamic range. Currently, however, an IP gateway interfacing with a class 5 switch cannot switch this 256 kbps wide-band voice traffic. The present inventors have thus concluded, based on the foregoing, that a need exists for improved methods and systems for switching this wide-band voice traffic.